-- Added extension '2' priority 1 to demo
-- Added extension '2' priority 2 to demo
-- Added extension '3' priority 1 to demo
-- Added extension '3' priority 2 to demo
-- Added extension '1000' priority 1 to demo
-- Added extension '1001' priority 1 to demo
-- Added extension '1234' priority 1 to demo
-- Added extension '1234' priority 2 to demo
-- Added extension '1235' priority 1 to demo
-- Added extension '1236' priority 1 to demo
-- Added extension '1236' priority 2 to demo
-- Added extension '#' priority 1 to demo
-- Added extension '#' priority 2 to demo
-- Added extension 't' priority 1 to demo
-- Added extension 'i' priority 1 to demo
-- Added extension '500' priority 1 to demo
-- Added extension '500' priority 2 to demo
-- Added extension '500' priority 3 to demo
-- Added extension '500' priority 4 to demo
-- Added extension '600' priority 1 to demo
-- Added extension '600' priority 2 to demo
-- Added extension '600' priority 3 to demo
-- Added extension '600' priority 4 to demo
-- Added extension '8500' priority 1 to demo
-- Added extension '8500' priority 2 to demo
-- Registered extension context 'default'
-- Including context 'demo' in context 'default'
-- Added extension '8500' priority 1 to default
-- Added extension '8500' priority 2 to default
-- Added extension '8600' priority 1 to default
-- Registered extension context 'sip'
-- Added extension '101' priority 1 to sip
-- Added extension '102' priority 1 to sip
-- Added extension '103' priority 1 to sip
-- Added extension '104' priority 1 to sip
-- Added extension '105' priority 1 to sip
-- Added extension '106' priority 1 to sip
-- Added extension '_X.' priority 1 to sip
-- Added extension '8500' priority 1 to sip
-- Added extension '8500' priority 2 to sip
-- Added extension '8600' priority 1 to sip
-- Added extension '_NXXXXXX' priority 1 to sip
-- Added extension '_1NXXNXXXXXX' priority 1 to sip
-- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Reloading module 'app_queue.so' (True Call Queueing)
== Parsing '/etc/asterisk/queues.conf': Found
-- Reloading module 'app_enumlookup.so' (ENUM Lookup)
== Parsing '/etc/asterisk/enum.conf': Found
-- Reloading module 'app_txtcidname.so' (TXTCIDName)
== Parsing '/etc/asterisk/enum.conf': Found
-- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend)
-- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend)
== Parsing '/etc/asterisk/cdr_manager.conf': Found
-- Reloading module 'cdr_odbc.so' (ODBC CDR Backend)
== Unregistered 'ODBC' CDR backend
== Parsing '/etc/asterisk/cdr_odbc.conf': Found
-- Reloading module 'cdr_pgsql.so' (PostgreSQL CDR Backend)
== Unregistered 'pgsql' CDR backend
== Parsing '/etc/asterisk/cdr_pgsql.conf': Found
Oct 22 16:04:20 ERROR[1076298368]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Un
able to connect to database server localhost. Calls will not be logged!
Oct 22 16:04:20 ERROR[1076298368]: cdr_pgsql.c:300 my_load_module: cdr_pgsql: Re
ason: could not connect to server: Connection refused
Is the server running on host localhost and accepting
TCP/IP connections on port 3306?
Reloading MGCP
== Parsing '/etc/asterisk/mgcp.conf': Found
== MGCP Listening on 0.0.0.0:2727
== Using TOS bits 0
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
-- Executing Dial("SIP/Phone2-a272", "Sip/g1/8178290827") in new stack
Oct 22 16:05:06 WARNING[1118864576]: chan_sip.c:1375 create_addr: No such host:
g1
Oct 22 16:05:06 NOTICE[1118864576]: app_dial.c:742 dial_exec: Unable to create c
hannel of type 'Sip'
== Everyone is busy/congested at this time
Oct 22 16:05:16 WARNING[1118864576]: pbx.c:1933 ast_pbx_run: Timeout, but no rul
e 't' in context 'sip'
Oct 22 16:05:16 ERROR[1118864576]: cdr_pgsql.c:66 pgsql_log: cdr_pgsql: Unable t
o connect to database server localhost. Calls will not be logged!
Oct 22 16:05:16 ERROR[1118864576]: cdr_pgsql.c:67 pgsql_log: cdr_pgsql: Reason:
could not connect to server: Connection refused
Is the server running on host localhost and accepting
TCP/IP connections on port 3306?
reload
== Parsing '/etc/asterisk/extconfig.conf': Found
== Parsing '/etc/asterisk/manager.conf': Found
Oct 22 16:11:21 NOTICE[1076298368]: manager.c:1437 init_manager: Use of portno i
n manager.conf deprecated. Please use 'port=5038' instead.
== Parsing '/etc/asterisk/enum.conf': Found
== Parsing '/etc/asterisk/rtp.conf': Found
== RTP Allocating from port range 10000 -> 20000
-- Reloading module 'cdr_addon_mysql.so' (MySQL CDR Backend)
== Unregistered 'mysql' CDR backend
== Parsing '/etc/asterisk/cdr_mysql.conf': Not found (No such file or director
y)
Oct 22 16:11:21 WARNING[1076298368]: cdr_addon_mysql.c:264 my_load_module: Unabl
e to load config for mysql CDR's: cdr_mysql.conf
-- Reloading module 'res_adsi.so' (ADSI Resource)
== Parsing '/etc/asterisk/adsi.conf': Found
-- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)
-- Reloading module 'res_indications.so' (Indications Configuration)
-- Unregistered indication country 'cl'
-- Unregistered indication country 'tw'
Oct 22 16:11:21 NOTICE[1076298368]: indications.c:397 ast_unregister_indication_
country: Removed default indication country 'us'
-- Unregistered indication country 'us'
-- Unregistered indication country 'au'
-- Unregistered indication country 'fr'
-- Unregistered indication country 'de'
-- Unregistered indication country 'nl'
-- Unregistered indication country 'uk'
-- Unregistered indication country 'fi'
-- Unregistered indication country 'no'
-- Unregistered indication country 'br'
-- Unregistered indication country 'za'
-- Unregistered indication country 'it'
-- Unregistered indication country 'us-o'
-- Unregistered indication country 'gr'
-- Unregistered indication country 'ru'
-- Unregistered indication country 'nz'
== Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'cl'
-- Registered indication country 'tw'
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Registered indication country 'fi'
-- Registered indication country 'no'
-- Registered indication country 'br'
-- Registered indication country 'za'
-- Registered indication country 'it'
-- Registered indication country 'us-o'
-- Registered indication country 'gr'
-- Registered indication country 'ru'
-- Registered indication country 'nz'
-- Setting default indication country to 'us'
-- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
-- Reloading module 'chan_agent.so' (Agent Proxy Channel)
== Parsing '/etc/asterisk/agents.conf': Found
-- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP))
-- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))
== Parsing '/etc/asterisk/iax.conf': Found
== Loaded firmware 'iaxy.bin'
== Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
-- Reloading module 'chan_local.so' (Local Proxy Channel)
-- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny
))
-- Reloading module 'pbx_config.so' (Text Extension Configuration)
== Parsing '/etc/asterisk/extensions.conf': Found
-- Setting global variable 'CONSOLE' to 'Console/dsp'
-- Setting global variable 'IAXINFO' to 'guest'
-- Setting global variable 'TRUNK' to 'Zap/g1'
-- Setting global variable 'TRUNKMSD' to '1'
-- Registered extension context 'iaxtel700'
-- Added extension '_91700XXXXXXX' priority 1 to iaxtel700
-- Registered extension context 'iaxprovider'
-- Registered extension context 'trunkint'
-- Added extension '_9011.' priority 1 to trunkint
-- Added extension '_9011.' priority 2 to trunkint
-- Registered extension context 'trunkld'
-- Added extension '_91NXXNXXXXXX' priority 1 to trunkld
-- Added extension '_91NXXNXXXXXX' priority 2 to trunkld
-- Registered extension context 'trunklocal'
-- Added extension '_9NXXXXXX' priority 1 to trunklocal
-- Added extension '_9NXXXXXX' priority 2 to trunklocal
-- Registered extension context 'trunktollfree'
-- Added extension '_91800NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91800NXXXXXX' priority 2 to trunktollfree
-- Added extension '_91888NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91888NXXXXXX' priority 2 to trunktollfree
-- Added extension '_91877NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91877NXXXXXX' priority 2 to trunktollfree
-- Added extension '_91866NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91866NXXXXXX' priority 2 to trunktollfree
-- Registered extension context 'international'
-- Including context 'longdistance' in context 'international'
-- Including context 'trunkint' in context 'international'
-- Registered extension context 'longdistance'
-- Including context 'local' in context 'longdistance'
-- Including context 'trunkld' in context 'longdistance'
-- Registered extension context 'local'
-- Including context 'default' in context 'local'
-- Including context 'parkedcalls' in context 'local'
-- Including context 'trunklocal' in context 'local'
-- Including context 'iaxtel700' in context 'local'
-- Including context 'trunktollfree' in context 'local'
-- Including context 'iaxprovider' in context 'local'
-- Registered extension context 'macro-stdexten'
-- Added extension 's' priority 1 to macro-stdexten
-- Added extension 's' priority 2 to macro-stdexten
-- Added extension 's-NOANSWER' priority 1 to macro-stdexten
-- Added extension 's-NOANSWER' priority 2 to macro-stdexten
-- Added extension 's-BUSY' priority 1 to macro-stdexten
-- Added extension 's-BUSY' priority 2 to macro-stdexten
-- Added extension '_s-.' priority 1 to macro-stdexten
-- Added extension 'a' priority 1 to macro-stdexten
-- Registered extension context 'demo'
-- Including context 'sip' in context 'demo'
-- Added extension 's' priority 1 to demo
-- Added extension 's' priority 2 to demo
-- Added extension 's' priority 3 to demo
-- Added extension 's' priority 4 to demo
-- Added extension 's' priority 5 to demo
-- Added extension 's' priority 6 to demo
-- Added extension '2' priority 1 to demo
-- Added extension '2' priority 2 to demo
-- Added extension '3' priority 1 to demo
-- Added extension '3' priority 2 to demo
-- Added extension '1000' priority 1 to demo
-- Added extension '1001' priority 1 to demo
-- Added extension '1234' priority 1 to demo
-- Added extension '1234' priority 2 to demo
-- Added extension '1235' priority 1 to demo
-- Added extension '1236' priority 1 to demo
-- Added extension '1236' priority 2 to demo
-- Added extension '#' priority 1 to demo
-- Added extension '#' priority 2 to demo
-- Added extension 't' priority 1 to demo
-- Added extension 'i' priority 1 to demo
-- Added extension '500' priority 1 to demo
-- Added extension '500' priority 2 to demo
-- Added extension '500' priority 3 to demo
-- Added extension '500' priority 4 to demo
-- Added extension '600' priority 1 to demo
-- Added extension '600' priority 2 to demo
-- Added extension '600' priority 3 to demo
-- Added extension '600' priority 4 to demo
-- Added extension '8500' priority 1 to demo
-- Added extension '8500' priority 2 to demo
-- Registered extension context 'default'
-- Including context 'demo' in context 'default'
-- Added extension '8500' priority 1 to default
-- Added extension '8500' priority 2 to default
-- Added extension '8600' priority 1 to default
-- Registered extension context 'sip'
-- Added extension '101' priority 1 to sip
-- Added extension '102' priority 1 to sip
-- Added extension '103' priority 1 to sip
-- Added extension '104' priority 1 to sip
-- Added extension '105' priority 1 to sip
-- Added extension '106' priority 1 to sip
-- Added extension '_X.' priority 1 to sip
-- Added extension '8500' priority 1 to sip
-- Added extension '8500' priority 2 to sip
-- Added extension '8600' priority 1 to sip
-- Added extension '_NXXXXXX' priority 1 to sip
-- Added extension '_1NXXNXXXXXX' priority 1 to sip
-- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Reloading module 'app_queue.so' (True Call Queueing)
== Parsing '/etc/asterisk/queues.conf': Found
-- Reloading module 'app_enumlookup.so' (ENUM Lookup)
== Parsing '/etc/asterisk/enum.conf': Found
-- Reloading module 'app_txtcidname.so' (TXTCIDName)
== Parsing '/etc/asterisk/enum.conf': Found
-- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend)
-- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend)
== Parsing '/etc/asterisk/cdr_manager.conf': Found
-- Reloading module 'cdr_odbc.so' (ODBC CDR Backend)
== Unregistered 'ODBC' CDR backend
== Parsing '/etc/asterisk/cdr_odbc.conf': Found
-- Reloading module 'cdr_pgsql.so' (PostgreSQL CDR Backend)
== Unregistered 'pgsql' CDR backend
== Parsing '/etc/asterisk/cdr_pgsql.conf': Found
Oct 22 16:11:21 ERROR[1076298368]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Un
able to connect to database server localhost. Calls will not be logged!
Oct 22 16:11:21 ERROR[1076298368]: cdr_pgsql.c:300 my_load_module: cdr_pgsql: Re
ason: could not connect to server: Connection refused
Is the server running on host localhost and accepting
TCP/IP connections on port 3306?
Reloading MGCP
== Parsing '/etc/asterisk/mgcp.conf': Found
== MGCP Listening on 0.0.0.0:2727
== Using TOS bits 0
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
-- Executing Dial("SIP/Phone2-032e", "Zap/g1/8178290827") in new stack
Oct 22 16:11:40 NOTICE[1118864576]: app_dial.c:742 dial_exec: Unable to create c
hannel of type 'Zap'
== Everyone is busy/congested at this time
Oct 22 16:11:50 WARNING[1118864576]: pbx.c:1933 ast_pbx_run: Timeout, but no rul
e 't' in context 'sip'
Oct 22 16:11:50 ERROR[1118864576]: cdr_pgsql.c:66 pgsql_log: cdr_pgsql: Unable t
o connect to database server localhost. Calls will not be logged!
Oct 22 16:11:50 ERROR[1118864576]: cdr_pgsql.c:67 pgsql_log: cdr_pgsql: Reason:
could not connect to server: Connection refused
Is the server running on host localhost and accepting
TCP/IP connections on port 3306?
Chan Extension Context Language MusicOnHol
pseudo sip
1 sip
2 sip
3 sip
4 sip
5 sip
6 sip
7 sip
8 sip
9 sip
10 sip
11 sip
12 sip
13 sip
14 sip
15 sip
16 sip
17 sip
18 sip
19 sip
20 sip
21 sip
22 sip
23 sip
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
; group => <trunkgroup>,<dchannel>[,<backup1>...]
;
; trunkgroup is the numerical trunk group to create
; dchannel is the zap channel which will have the
; d-channel for the trunk.
; backup1 is an optional list of backup d-channels.
;
;trunkgroup => 1,24,48
;
; Spanmap: Associates a span with a trunk group
; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
;
; zapspan is the zap span number to associate
; trunkgroup is the trunkgroup (specified above) for the mapping
; logicalspan is the logical span number within the trunk group to use.
; if unspecified, no logical span number is used.
;
;spanmap => 1,1,1
;spanmap => 2,1,2
;spanmap => 3,1,3
;spanmap => 4,1,4
[channels]
;
; Default language
;
;language=en
;
; Default context
;
switchtype=national
context=sip
signalling=pri_cpe
group=1
channel => 1-23
;
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN
; ni1: Old National ISDN 1
;
switchtype=national
context=sip
signalling=pri_cpe
group=1
channel => 1-23
;
; Some switches (AT&T especially) require network specific facility IE
; supported values are currently 'none', 'sdn', 'megacom', 'accunet'
;
;nsf=none
;
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;pridialplan=national
;
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan)
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
;prilocaldialplan=national
;
; Overlap dialing mode (sending overlap digits)
;
;overlapdial=yes
;
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
;
; priindication = outofband
;
;
; Signalling method (default is fxs). Valid values:
; em: E & M
; em_w: E & M Wink
; featd: Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featb: Feature Group B (MF (domestic, US))
; fxs_ls: FXS (Loop Start)
; fxs_gs: FXS (Ground Start)
; fxs_ks: FXS (Kewl Start)
; fxo_ls: FXO (Loop Start)
; fxo_gs: FXO (Ground Start)
; fxo_ks: FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
; sf: SF (Inband Tone) Signalling
; sf_w: SF Wink
; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
; sf_featb: SF Feature Group B (MF (domestic, US))
; The following are used for Radio interfaces:
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank)
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank)
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the channel bank)
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank)
; em_rx: Receive audio/COR on an E&M interface (1-way)
; em_tx: Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface (2-way)
; em_rxtx: same as em_txrx (for our dyslexic friends)
; sf_rx: Receive audio/COR on an SF interface (1-way)
; sf_tx: Transmit audio/PTT on an SF interface (1-way)
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface (2-way)
; sf_rxtx: same as sf_txrx (for our dyslexic friends)
;
;signalling=fxo_ls
;
; A variety of timing parameters can be specified as well
; Including:
; prewink: Pre-wink time (default 50ms)
; preflash: Pre-flash time (default 50ms)
; wink: Wink time (default 150ms)
; flash: Flash time (default 750ms)
; start: Start time (default 1500ms)
; rxwink: Receiver wink time (default 300ms)
; rxflash: Receiver flashtime (default 1250ms)
; debounce: Debounce timing (default 600ms)
;
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
;
; Whether or not to use caller ID
;
usecallerid=yes
;
; Type of caller ID signalling in use
; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands
;
;cidsignalling=bell
;
; What signals the start of caller ID
; ring = a ring signals the start, polarity = polarity reversal signals the start
;
;cidstart=ring
;
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
;
hidecallerid=no
;
; Whether or not to enable call waiting on FXO lines
;
callwaiting=yes
;
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user)
; Mostly use with FXS ports
;
;restrictcid=no
;
; Whether or not use the caller ID presentation for the outgoing call that the calling switch is sending
;
usecallingpres=yes
;
; Support Caller*ID on Call Waiting
;
callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes
;
; Support call forward variable
;
cancallforward=yes
;
; Whether or not to support Call Return (*69)
;
callreturn=yes
;
; Stutter dialtone support: If a mailbox is specified without a voicemail
; context, then when voicemail is received in a mailbox in the default
; voicemail context in voicemail.conf, taking the phone off hook will
; cause a stutter dialtone instead of a normal one.
;
; If a mailbox is specified *with* a voicemail context, the same will
; result if voicemail recieved in mailbox in the specified voicemail
; context
;
; for default voicemail context, the example below is fine:
;
;mailbox=1234
;
; for any other voicemail context, the following will produce the
; stutter tone:
;
;mailbox=1234@context
;
; Enable echo cancellation
; Use either "yes", "no", or a power of two from 32 to 256 if you wish
; to actually set the number of taps of cancellation.
;
echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel
; when the circuit path is entirely TDM. You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
;
echocancelwhenbridged=yes
;
; In some cases, the echo canceller doesn't train quickly enough and there
; is echo at the beginning of the call. Enabling echo training will cause
; asterisk to briefly mute the channel, send an impulse, and use the impulse
; response to pre-train the echo canceller so it can start out with a much
; closer idea of the actual echo. Value may be "yes", "no", or a number of
; milliseconds to delay before training (default = 400)
;
;echotraining=yes
;echotraining=800
;
; If you are having trouble with DTMF detection, you can relax the
; DTMF detection parameters. Relaxing them may make the DTMF detector
; more likely to have "talkoff" where DTMF is detected when it
; shouldn't be.
;
;relaxdtmf=yes
;
; You may also set the default receive and transmit gains (in dB)
;
rxgain=0.0
txgain=0.0
;
; Logical groups can be assigned to allow outgoing rollover. Groups
; range from 0 to 31, and multiple groups can be specified.
;
group=1
;
; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#. For simple offices, just
; make these both the same
;
callgroup=1
pickupgroup=1
;
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
immediate=no
;
; CallerID can be set to "asreceived" or a specific number
; if you want to override it. Note that "asreceived" only
; applies to trunk interfaces.
;
;callerid=2564286000
;
; AMA flags affects the recording of Call Detail Records. If specified
; it may be 'default', 'omit', 'billing', or 'documentation'.
;
;amaflags=default
;
; Channels may be associated with an account code to ease
; billing
;
;accountcode=lss0101
;
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
;
;adsi=yes
;
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
; etc, it can be useful to perform busy detection either in an effort to
; detect hangup or for detecting busies
;
;busydetect=yes
;
; If busydetect is enabled, is also possible to specify how many
; busy tones to wait before hanging up. The default is 4, but
; better results can be achieved if set to 6 or even 8. Mind that
; higher the number, more time is needed to hangup a channel, but
; lower is probability to get random hangups
;
;busycount=4
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate.
;
; Few zones are supported at the time of this writing, but may
; be selected with "progzone"
;
; This feature can also easily detect false hangups. The symptoms of this
; is being disconnected in the middle of a call for no reason.
;
;callprogress=yes
;progzone=us
;
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
;
;pulsedial=yes
;
; For fax detection, uncomment one of the following lines. The default is *OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;
; Select which class of music to use for music on hold. If not specified
; then the default will be used.
;
;musiconhold=default
;
; PRI channels can have an idle extension and a minunused number. So long
; as at least "minunused" channels are idle, chan_zap will try to call
; "idledial" on them, and then dump them into the PBX in the "idleext"
; extension (which is of the form exten@context). When channels are needed
; the "idle" calls are disconnected (so long as there are at least "minidle"
; calls still running, of course) to make more channels available. The
; primary use of this is to create a dynamic service, where idle channels
; are bundled through multilink PPP, thus more efficiently utilizing
; combined voice/data services than conventional fixed mappings/muxings.
;
;idledial=6999
;idleext=6999@dialout
;minunused=2
;minidle=1
;
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
;jitterbuffers=4
;
; You can define your own custom ring cadences here. You can define up to
; 8 pairs. If the silence is negative, it indicates where the callerid
; spill is to be placed. Also, if you define any custom cadences, the
; default cadences will be turned off.
;
; Syntax is: cadence=ring,silence[,ring,silence[...]]
;
; These are the default cadences:
;
;cadence=125,125,2000,-4000
;cadence=250,250,500,1000,250,250,500,-4000
;cadence=125,125,125,125,125,-4000
;cadence=1000,500,2500,-5000
;
; Each channel consists of the channel number or range. It
; inherits the parameters that were specified above its declaration
;
; For GR-303, CRV's are created like channels except they must start
; with the trunk group followed by a colon, e.g.:
;
; crv => 1:1
; crv => 2:1-2,5-8
;
;
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
;
; For example, maybe we have some other channels
; which start out in a different context and use
; E & M signalling instead.
;
;context=remote
;sigalling=em
;channel => 15
;channel => 16
;signalling=em_w
;
; All those in group 0 I'll use for outgoing calls
;
; Strip most significant digit (9) before sending
;
;stripmsd=1
;callerid=asreceived
;group=0
;signalling=fxs_ls
;channel => 45
;signalling=fxo_ls
;group=1
;callerid="Joe Schmoe" <(256) 428-6131>
;channel => 25
;callerid="Megan May" <(256) 428-6132>
;channel => 26
;callerid="Suzy Queue" <(256) 428-6233>
;channel => 27
;callerid="Larry Moe" <(256) 428-6234>
;channel => 28
;
; Sample PRI (CPE) config: Specify the switchtype, the signalling as
; either pri_cpe or pri_net for CPE or Network termination, and generally
; you will want to create a single "group" for all channels of the PRI.
;
; switchtype = national
; signalling = pri_cpe
; group = 2
; channel => 1-23
;
; Used for distintive ring support for x100p.
; You can see the dringX patterns is to set any one of the dringXcontext fields
; and they will be printed on the console when an inbound call comes in.
;
;dring1=95,0,0
;dring1context=internal1
;dring2=325,95,0
;dring2context=internal2
; If no pattern is matched here is where we go.
;context=default
;channel => 1
Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=<span num>,<timing>,<line build out (LBO)>,<framing>,<coding>[,yellow]
#
# The timing parameter determines the selection of primary, secondary, and
# so on sync sources. If this span should be considered a primary sync
# source, then give it a value of "1". For a secondary, use "2", and so on.
# To not use this as a sync source, just use "0"
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1
#
# Note: "d4" could be referred to as "sf" or "superframe"
#
# The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1
#
# E1's may have the additional keyword "crc4" to enable CRC4 checking
#
# If the keyword "yellow" follows, yellow alarm is transmitted when no
# channels are open.
#
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=<driver>,<address>,<numchans>,<timing>
#
# Where <driver> is the name of the driver (e.g. eth), <address> is the
# driver specific address (like a MAC for eth), <numchans> is the number
# of channels, and <timing> is a timing priority, like for a normal span.
# use "0" to not use this as a timing source, or prioritize them as
# primary, secondard, etc. Note that you MUST have a REAL zaptel device
# if you are not using external timing.
#
# dynamic=eth,eth0/00:02:b3:35:43:9c,24,0
#
# Next come the definitions for using the channels. The format is:
# <device>=<channel list>
#
# Valid devices are:
#
# "e&m" : Channel(s) are signalled using E&M signalling (specific
# implementation, such as Immediate, Wink, or Feature Group D
# are handled by the userspace library).
# "fxsls" : Channel(s) are signalled using FXS Loopstart protocol.
# "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol.
# "fxsks" : Channel(s) are signalled using FXS Koolstart protocol.
# "fxols" : Channel(s) are signalled using FXO Loopstart protocol.
# "fxogs" : Channel(s) are signalled using FXO Groundstart protocol.
# "fxoks" : Channel(s) are signalled using FXO Koolstart protocol.
# "sf" : Channel(s) are signalled using in-band single freq tone.
# Syntax as follows:
# channel# => sf:<rxfreq>,<rxbw>,<rxflag>,<txfreq>,<txlevel>,<txflag>
# rxfreq is rx tone freq in hz, rxbw is rx notch (and decode)
# bandwith in hz (typically 10.0), rxflag is either 'normal' or
# 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone
# level in dbm, txflag is either 'normal' or 'inverted'. Set
# rxfreq or txfreq to 0.0 if that tone is not desired.
# "unused" : No signalling is performed, each channel in the list remains idle
# "clear" : Channel(s) are bundled into a single span. No conversion or
# signalling is performed, and raw data is available on the master.
# "indclear": Like "clear" except all channels are treated individually and
# are not bundled. "bchan" is an alias for this.
# "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the
# bundle, and the resulting data is communicated via the master
# device.
# "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the
# bundle and also performs incoming and outgoing FCS insertion
# and verification. "dchan" is an alias for this.
# "nethdlc" : The zaptel driver bundles the channels together into an
# hdlc network device, which in turn can be configured with
# sethdlc (available separately).
# "dacs" : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
# "dacsrbs" : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon and
# also performs the DACSing of RBS bits
#
# The channel list is a comma-separated list of channels or ranges, for
# example:
#
# 1,3,5 (channels one, three, and five)
# 16-23, 29 (channels 16 through 23, as well as channel 29
#
# So, some complete examples are:
# e&m=1-12
# nethdlc=13-24
# fxsls=25,26,27,28
# fxols=29-32
#
#fxoks=1-24
#bchan=25-47
#dchan=48
#fxols=1-12
#fxols=13-24
#e&m=25-29
#nethdlc=30-33
#clear=44
#clear=45
#clear=46
#clear=47
#fcshdlc=48
#dacs=1-24:48
#dacsrbs=1-24:48
#
# Finally, you can preload some tone zones, to prevent them from getting
# overwritten by other users (if you allow non-root users to open /dev/zap/*
# interfaces anyway. Also this means they won't have to be loaded at runtime.
# The format is "loadzone=<zone>" where the zone is a two letter country code.
#
# You may also specify a default zone with "defaultzone=<zone>" where zone
# is a two letter country code.
#
loadzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
defaultzone=us